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Topic: Sub-$100 shotgun? (Read 3187 times)

Anyone know of a sub-$100 phantom-powered hypercardioid with XLR out they can recommend? I won't be using it for music so it doesn't need to be great-sounding. In fact, it doesn't even need to be halfway decent-sounding for my application. Self-noise? Also not a problem; it just has to work. The hypercardioid version of the ECM8000 omni?

Hard to find a hypercardioid under $100 other than the Little Gem, but there are a number of very cheap shotguns with XLR out if you search the typical retailers. Why that pattern specifically if you don't care how it sounds?

It's an odd hybrid M-S arrangement that I figured out on my own though I bet I just re-invented an old wheel. The results are quite good.

I volunteer at our local community radio station as studio engineer and music festival remote broadcast producer. Earlier this month at Summer Festival here in Bend, Oregon, I tried this setup and am very pleased with how it sounds.

So, at these festivals (two more remaining this summer) we get a split from the PA system sound board, which is pretty dry and totally mono, because the sound mixer isn't interested in giving us a recording mix, he is only interested in how it sounds at FOH. So that's the feed we get, and we've been broadcasting that feed for years.

At this last festival, tried something new: I mixed that feed with an M-S mic arrangement I set up 40-or-so feet back in the audience. It works great.

To time-align the PA feed with the mic array -- the sound board signal gets to our broadcast/recording mixer before the sound hits the mics -- I have a delay fx box inline with the PA feed. I recorded the waveforms of the PA and the M mic during the first song, then stopped the recording and examined the PA and the M mic waveforms to determine how much the PA fed led the M mic signal, and set fx box delay to the appropriate amount. The waveforms line up, the sound is clear.

By then adding the S (side) mics to the mix I got very nice stereo spaciousness while remaining fully mono-compatible for FM broadcasting (the station is currently in mono for technical reasons).

If I was just recording the show, and not broadcasting it live, I could take the PA and mic tracks home then time-align them in my DAW and mix it down. That would be easy. But I have to dial in the delay quickly because we are broadcasting live, and for each festival the mic location will be different, and there's no predicting how much latency the PA board adds because they use different boards all the time. So I will always have to perform this initial calibration process, and I want to be able to do it quickly.

So I'm thinking that having a clearer waveform from the M mic, with less "room" reverb, would speed the process.

The cardioid I used this last time for the M mic picked up quite a bit of "room," so its waveform, compared with the PA feed, is pretty blurred and it takes a while to find an easily-identifiable sharp match between what I'm getting from the board and what the mic picks up. I am thinking that with a shotgun-type mic I'd get less room sound and thus a cleaner signal to use to compare with the board feed, making it easier to quickly dial in the delay. That's my swell idea, anyhow.

Why don't I care how the mic sounds? Because it turns out that once time-aligned, I don't need the M mic in the mix -- the sound board feed sits quite nicely in the middle, and the M mic contributes nothing really useful to the final mix. So that mic is really going to be used for calibration purposes, not for final audio.

I hope this makes sense. Shotgun, phantom powered, under $100, sound quality not so important. As you say, there are plenty of sub-$100 short shotguns on the market for video guys, so yeah, maybe one of those.

That's quite an interesting situation, and it sounds like you've come up with very creative solutions to your challenges.

I'm wondering if you could possibly simplify this setup and eliminate the delay box by adding a second fig-8 to make a Blumlein array, placed much closer to the stage. Depending on how good the PA mix is, you could then flip the direct / aud balance around compared to how you're currently doing this, making more like a classical recording setup whereby the Blumlein is your main pickup and the board feed are treated like solo spot mics, down in level compared to the main stereo pickup. You'd still have your mono compatibility this way, but then you would have a bit more control over the main part of the sound being sent out for broadcast.

If there aren't many channels for the PA, any chance of the FOH engineer giving you separate channel sub-outs that you could run to your own mixer? (I mean, we can dream, can't we?)

All of this involves spending way more than $100 for a shotgun mic, I know. Just spitballing ideas...

"If there aren't many channels for the PA, any chance of the FOH engineer giving you separate channel sub-outs that you could run to your own mixer? (I mean, we can dream, can't we?)"

That would be very dependent on the sound guy having a mixer with all them splits, me having a nice expensive snake of up to 100 feet, me springing for a much fancier board than I have, me learning how to mix live multitrack sound, and me having a lot more time on my hands than I have what with wrangling on-air hosts, queuing up underwriter spots, monitoring the broadcast signal, and all the other fun stuff that goes with broadcasting these events. Nah, ain't gonna happen. But thanks for the suggestion!

"If there aren't many channels for the PA, any chance of the FOH engineer giving you separate channel sub-outs that you could run to your own mixer? (I mean, we can dream, can't we?)"

That would be very dependent on the sound guy having a mixer with all them splits, me having a nice expensive snake of up to 100 feet, me springing for a much fancier board than I have, me learning how to mix live multitrack sound, and me having a lot more time on my hands than I have what with wrangling on-air hosts, queuing up underwriter spots, monitoring the broadcast signal, and all the other fun stuff that goes with broadcasting these events. Nah, ain't gonna happen. But thanks for the suggestion!

I get the picture... Just trying to save you work, not create more. Good luck with your setup!

You bet. kpov.org -- our next festival will be Sept 30, time of broadcast yet to be determined. We're talking USA, Pacific coast time, probably starting around noon and until maybe 10pm. This will be a very challenging show as our booth will be located midway between two stages, and when the act on one stage finishes, the act on the other will start. So I'll be scrambling in that gap to switch sound board feeds and change the micing. My plan is to set up the M-S array midway between the two stages and as well-aligned as we can make it so the distance from the mic to each stage's left-right speakers are as close as possible. Of course the stage riggers aren't going to worry about us to what might be the right spot for Stage A could be 30 feet away from the right spot for Stage B. There will be two boards, as well, and unlikely they will be the same make/model so the board latency from one board will be different than that from the other. My hope is that on Friday night I can tinker with that shotgun mic, point it at the first act on Stage A, determine the time offset between that stage's board feed and the arrival of the sound at the mic array, make a note of it, then when Stage B start up, turn the shotgun around to point at that stage and determine my delay setting for that stage. The S mic, being a figure 8, should not need to be moved. The stand for it will be staked to the ground.

This will be the "Bend Roots Revival" music festival and the bands are local, many amateur, some are just kids. The sound systems are a catch-as-catch-can affair, too. So no promises about the quality of the music or the sound. I'll be lucky to get board feeds from both stages. I'll be on IRC channel kpov if you want to comment. But be kind because by the time you hear that stream it will have been passed through our mixer at the remote, then through a peak limiter, then digitized to 16 bits / 48kHz, encoded to 192kbps ogg, transported to the station via Internet, played back on a Mac Mini in the studio using an Echo sound card into a 1980s-vintage opamp-based console, run through another peak limiter (to keep the signal legal and prevent clipping the station's live digital stream), re-encoded to 16/48 again, and sent to our stream provider as 192kbps mp3. Frankly with all that going on I'm surprised it even sounds like anything resembling music.

This will be a very challenging show as our booth will be located midway between two stages, and when the act on one stage finishes, the act on the other will start.

I was in this situation once, only it was indoors, and there was only one SBD involved. We did not even bother with the SBD at those stages because of the hassle. We used spaced omnis, as there was not even enough time to spin the mics. It was a challenge just to start a new track on my laptop between sets. The year before, at a different location, the stages were about a football field away from each other. A central mic setup would have sounded like ass, regardless of having separate SBD feeds, and there was almost a minute between sets, max.

IMHO, you would be better off with two rigs, and just switching witch one is broadcasting.

"IMHO, you would be better off with two rigs, and just switching witch one is broadcasting. "

You may well be right. The "rig" (mics, tall lighting stand, cables) are being bought out of my own pocket, the station as zero money for such things. Since this two-stages setup happens only once a year, I can't personally justify the expense. If it works out okay, I'll stick with it in 2018. If it sounds like doo-doo then maybe next year I'll save up Social Security pennies and but a second stand, shotgun and figure-8 mics, and build two more 100-foot cables. We shall see.

It's an odd hybrid M-S arrangement that I figured out on my own though I bet I just re-invented an old wheel. I mixed that feed with an M-S mic arrangement I set up 40-or-so feet back in the audience. It works great.

To time-align the PA feed with the mic array -- the sound board signal gets to our broadcast/recording mixer before the sound hits the mics -- I have a delay fx box inline with the PA feed. I recorded the waveforms of the PA and the M mic during the first song, then stopped the recording and examined the PA and the M mic waveforms to determine how much the PA fed led the M mic signal, and set fx box delay to the appropriate amount. The waveforms line up, the sound is clear.

By then adding the S (side) mics to the mix I got very nice stereo spaciousness while remaining fully mono-compatible for FM broadcasting (the station is currently in mono for technical reasons).

The cardioid I used this last time for the M mic picked up quite a bit of "room," so its waveform, compared with the PA feed, is pretty blurred and it takes a while to find an easily-identifiable sharp match between what I'm getting from the board and what the mic picks up. I am thinking that with a shotgun-type mic I'd get less room sound and thus a cleaner signal to use to compare with the board feed, making it easier to quickly dial in the delay. That's my swell idea, anyhow.

Why don't I care how the mic sounds? Because it turns out that once time-aligned, I don't need the M mic in the mix -- the sound board feed sits quite nicely in the middle, and the M mic contributes nothing really useful to the final mix. So that mic is really going to be used for calibration purposes, not for final audio.

This is great. I totally dig it. I've advocated similar approaches thinking about what audience microphone setup is most appropriate when a soundboard feed is available to the taper. The standard directional stereo microphone configurations pointing at the PA are providing a lot of redundant the direct sound information (the M component) compared to the SBD feed, and the stereo microphone configs used for audience recording are most often selected with Mid-component clarity as the top priority and things like stereo image width and spaciousness as nice to have, but secondary concerns. Yes, redundancy has value, yet what we really need from the AUD microphones in this situation is stereo width, reverberant spaciousness, ambience and audience reaction (the S component stuff). The SBD feed is providing all the M component clarity we need, in an even more clear and direct way.

Taken to the logical extreme, when the SBD is good sounding and reliable we really only need room mics to provide the best spatial capture of the room and audience ambience. Room mic'ing techniques become most appropriate: Wide spaced omnis or spaced bi-directionals facing sideways (Hamasaki Square like). If there is room ambience and audience behind the recording location, then it makes sense to use rearward facing spaced cardioids, or a standard near-spaced microphone configuration turned around and facing away from the stage. The intent becomes excluding the direct PA sound component as much as possible rather than optimizing for it's pickup, which allows for more useful ambient/Side signal from the audience mics. We can use more of that if it before we get too much overlapping Mid info. Much like professional live recording setups which often use supercards or shotguns on the outer edges of the stage facing the audience- those mics are intended to exclude pickup of the PA and on stage sound as much as possible, focusing exclusively on the room and audience as much as possible.

This is also how I think about multi-microphone AUD-only techiques these days, why I advocate either a single very-directional microphone in the center facing forward (as M-component / SBD surrogate) or a PAS X/Y pair for the same reason, combined with other mics pointing mostly sideways and backwards for stereo imaging and ambient S component pickup.

Please stick around TS! Your insights and inventive thinking will be valuable in some of the technical discussions around here which deal with this stuff.

Best of luck with the microphone search and the broadcasts. Agreed that the video short shotgun market is probably fertile ground with good low cost options. BTW, I use Naiant X-8S inexpensive bi-directionals as Side microphones. Good stuff.

"Please stick around TS! Your insights and inventive thinking will be valuable in some of the technical discussions around here which deal with this stuff."

Thank you!

"The SBD feed is providing all the M component clarity we need, in an even more clear and direct way."

And to address an idea proposed earlier in this thread, the SBD feed is in every way superior to what the PA speakers are putting out. Adding in speaker coloration never helps. I use active noise-cancelling headphones at these shows and I can hear the SBD very clearly. When I take off the cans, the sound coming from the stage is 'way crummier. Using the SBD the way I would use a spotlight mic in an orchestral recording was my original thinking but I've scrapped that what the M mic is hearing is nearly useless, sonically. It contributes only a blurred, muddier, sloppier version of the SBD. The M mic isn't even suitable for wetting the SBD because it sounds such a mess. Useless, IMO.

Except for one teentsy little exception. Last year we had a Led Zeppelin tribute band play. The guitar amp -- Marshall stack, natch -- was turned up so loud that the PA sound engineer didn't put any of it into the PA because the audience could hear the guitar quite well without it. I wasn't using any AUD mics. So the broadcast listeners heard NO guitar. "The PA is the inverse of the stage" is how a sound man I really admire put it. So this is a situation where you would want a live M mic rigged up to fill in what was left out of the PA mix.

On the last festival, being the first using this hybrid M/SBD-S mic setup, I did a rough mix of the SBD and the S mics for broadcast and left it at that. Next time there is something unbalanced like that Led Zeppelin tribute band a-playin' away onstage I will add in some M mic so the broadcast listeners will know there's someone on guitar.

I record everything separately on separate tracks for later mixdown and archiving: the SBD, the S mic, and the M mic. A little mixing, a little mastering, and Hey Presto! a not bad-sounding recording.

And to address an idea proposed earlier in this thread, the SBD feed is in every way superior to what the PA speakers are putting out. Adding in speaker coloration never helps. I use active noise-cancelling headphones at these shows and I can hear the SBD very clearly. When I take off the cans, the sound coming from the stage is 'way crummier. Using the SBD the way I would use a spotlight mic in an orchestral recording was my original thinking but I've scrapped that what the M mic is hearing is nearly useless, sonically. It contributes only a blurred, muddier, sloppier version of the SBD. The M mic isn't even suitable for wetting the SBD because it sounds such a mess. Useless, IMO.

Yes, and why I've been arguing for multichannel recording arrays that use a center microphone(s) intended to focus on the direct sound arriving from stage and PA to the exclusion of all ambient and hall sound as much as possible. Really, that is in many ways an attempt to emulate what one would get from a complete and well-balanced SBD feed as closely as possible when one is not available.

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Except for one teentsy little exception. Last year we had a Led Zeppelin tribute band play. The guitar amp -- Marshall stack, natch -- was turned up so loud that the PA sound engineer didn't put any of it into the PA because the audience could hear the guitar quite well without it. I wasn't using any AUD mics. So the broadcast listeners heard NO guitar. "The PA is the inverse of the stage" is how a sound man I really admire put it. So this is a situation where you would want a live M mic rigged up to fill in what was left out of the PA mix.

On the last festival, being the first using this hybrid M/SBD-S mic setup, I did a rough mix of the SBD and the S mics for broadcast and left it at that. Next time there is something unbalanced like that Led Zeppelin tribute band a-playin' away onstage I will add in some M mic so the broadcast listeners will know there's someone on guitar.

I record everything separately on separate tracks for later mixdown and archiving: the SBD, the S mic, and the M mic. A little mixing, a little mastering, and Hey Presto! a not bad-sounding recording.

That's a common problem with SBD feeds many tapers are familiar with, and somewhat of a realist counter point to my argument that all we really need along with a SBD feed is room mics, which I should clarify. A SBD PA feed is not always "complete and well-balanced". This is typically most noticeable with electric guitars and other highly amplified on-stage sources which are loud enough on their own that they are not reinforced significantly through the PA. It's a bigger issue at small clubs and smaller outdoor events where the PA is relatively small and works along with with the sound generated on-stage to properly illuminate the room. At very large events the PA dominates the on-stage sound and is mixed in a more "complete and well-balanced" way. The other less dramatic way this can sometimes show up is via whatever EQ filtering has been applied to best adapt the PA to the acoustical environment of the room (if your feed is tapped after that correction). If a funky EQ curve has been used to get a reasonable flat sounding result out in the room, that curve is basically the inverse of the room resonances. It works with the room's response to produce a relatively flat sounding end result for folks in room. But in your direct PA feed that curve is not flattened by the room response and you will hear the corrective curve instead of the room-averaged end result.

Two fixes for this-1) Get the FOH engineer to provide you with a dedicated stereo mix separate from the PA feed. That should have all instrumentation in it which passes through the board and should be free of any room EQ corrections, if you're lucky it will be nicely stereo panned as well. Problems are that this is may not be an option, and even if it is, getting that mix well balanced and keeping it that way is not going to be a focus of the FOH. Their job is managing the house sound out in the room.2) Mic the on-stage sound. This is best done with a mic or pair of mics on the stage or at the stage-lip. If you are lucky you can put an omni on stage, or a pair of mics at the stage-lip for this purpose and run them through the FOH snake back to the board for you to use, otherwise you'd need to run and manage long cables. This works great and picks up lots of detail, depth and imaging cues which never make out into the room to relatively distant AUD mics or through close-mics to the PA. Drum kits and acoustic instruments sound more open, clear, sparkly with 3-dimensional sound-staging and the up-front presence realism quotient goes way up.

With a SBD feed, on-stage mics, and audience mics for ambiance, pretty much all bases are covered. With traditional AUD recording we are trying to get all three of those aspect from back at the audience recording location, where only the ambient audience stuff is maybe correct, but even much of that stuff (the audience reaction for sure) is usually best captured closer to the front.

Your best answer would be to put your M/S pair at the front of the stage and use an omni Mid mic. That would make everything easier for you and should sound much better as well. No time alignment required. Use the SBD as primary Mid and mix in a bit of omni Mid with it to get some audience ambience into the mono broadcast and take care of any loud guitars which are not well represented in the SBD feed. Gets great stage presence and clarity plus the best, most energetic crowd reaction from up front rather than distracted talking further back. Stereoized with the addition of the Side mic you also get nice wide, deep, 3d-like on stage and audience imaging.

Can you talk the FOH into letting you use two channels of the snake to run your stage mics back to the board location? Otherwise it may be worthwhile to figure out a way to safely run long cables up there.

This might not be doable, but would solve a lot of problems and should sound great. If not right away, maybe you can plant some seeds to eventualy move things that direction.

"IMHO, you would be better off with two rigs, and just switching witch one is broadcasting. "

You may well be right. The "rig" (mics, tall lighting stand, cables) are being bought out of my own pocket, the station as zero money for such things. Since this two-stages setup happens only once a year, I can't personally justify the expense. If it works out okay, I'll stick with it in 2018. If it sounds like doo-doo then maybe next year I'll save up Social Security pennies and but a second stand, shotgun and figure-8 mics, and build two more 100-foot cables. We shall see.

Got to go with what you got. There was 5 or 6 of us on the team, so we had plenty of gear options to choose from. You should at least consider getting a friend to lend a hand, if that is possible, even if they are not a taper. That would take some of the load off of switching.

"Your best answer would be to put your M/S pair at the front of the stage and use an omni Mid mic."

You make a compelling argument. I think I'll spring for a second M-S rig so I can mic both stages. And more XLR cables, this time in all the colors of the rainbow so that by the time I get six cables back to our booth (2 each SBD, Mid mic, and Side mic) so I can tell them apart. I have just that many mic preamps available (one Focusrite Scarlett 18i8 and one 6i6) strapped via SPDIF.

You'd use an omni for the Mid mic? That's going to be pretty heavy in the bass and my preamps don't have any bass-cut controls. I have a couple of subcards I could use at the expense of rolling off off-axis highs.

This is going to be an ambitious project.

For the suggestion that I get help -- the station and I are constantly requesting for volunteers to aid in these events. One thing about community radio volunteers is that nearly none of them are technically-adept or have a background in mixing audio. The booth is already full up with me, the equipment, and two announcers. There isn't room for a fourth.

I figure that once I have my levels dialed in, I just need to mute three inputs and unmute the other three inputs to switch between stages. And, this being community radio and not some kind of fancy-pants operation where we are being paid and expected to crank out flawless sound, some fumbling is normal. This is Oregon, man. Are you familiar with the cannabis laws here?

An omni Mid mic at the front edge of the stage should provide a good balance between the on-stage sound and audience sound, and the front/back balance will not change depending on how much Side you use. You could use a subcardiod Mid, just decide which is more important: making sure you get clean and clear audience enthusiasm or enough level from any guitar amps not strongly represented in the SBD. And with a directional Mid microphone the front/back balance will change as you dial in less or more Side, but at least that's not as pronounced as it would be if using a cardioid Mid.

You might try it with the subcard pointed one way, then the other to determine what works best. Without being there and seeing the setup, I'd probably choose to point the subcard out at the audience it is not in the SBD at all, providing better audience presence for all acts, and I suspect occasional act with loud guitar amps will illuminate the entire stage and near audience with plenty of reflected guitar sound so I doubt that would be lacking in a subcard Mid pointed away from the stage. And with some Side mic mixed in there should be even more on-stage guitar. Alternately, with the Subcard pointed at the stage you get nuanced on-stage depth and detail lacking in the SBD. Probably a call best made on the spot, after trying it both ways.

Since you don't have EQ control, it may depend on on how bass heavy it is up there. If its a thick subwoofer swamp at the front of the stage, then you may need to avoid the omni Mid unless you have a low-cut on the microphone itself you can engage. A subcard will be a bit less low bass sensitive which may help, but probably won't cut the bass by a huge amount.

I figure that once I have my levels dialed in, I just need to mute three inputs and unmute the other three inputs to switch between stages. And, this being community radio and not some kind of fancy-pants operation where we are being paid and expected to crank out flawless sound, some fumbling is normal. This is Oregon, man. Are you familiar with the cannabis laws here?

Heading out to Idaho for the eclipse next month, rafting down the Snake just prior, so will be right on the other side of your Eastern boarder. Rather tempted to drive over to the other side for the delve into darkness.

I already have four omni mics in my kit. None have low-cut switches. Does such a thing exist in the sub-$100 range with not-awful sound quality?

I'll also look at XLR inline high-pass filters, maybe build a couple. I have in inquiry in with Focusrite about the mic input impedance for the 1st Gen Scarlett 6i6 and 18i8. Doesn't appear to be part of the published specifications.

Using in-line high pass filters on the omni Mids is a wise idea as long as the component values are correct and it sounds like you have a handle on that. Even if not overly bass heavy at the mic location their use shouldn't be problematic as the SBD feed will provide low end extension. You shouldn't need them on the bi-directionals.

I figure I can pretty easily build a pair of second-order LC high-pass filters with some good film caps and inductors from the likes of Digikey. Put them inside a $33 eBay two-channel phantom power supply box with the guts removed. Where would you put the -3dB point for this kind of between-the-subwoofers omni micing situation.

I've pretty much decided to set my M-S rig at the front of the stage, and use an omni for the Mid mic. Bass is likely to be pretty heavy thereabouts, and I'm going to build a low-cut filter to use inline with the mic cable.

I can't find information on how steeply the average cardioid mic rolls off bass when proximity effect is not a factor . . . I'm guessing 6dB/octave (one-pole) but don't know. I suppose my filter should try to mimic the low-end response of a cardioid in the farfield. But steeper might be better.

So, what would you suggest for the low-cut filter design: one- or two-pole (6dB/oct or 12dB/oct), and where would you suggest I put my -3dB point.

Hmm. Good question. Various cardioids roll off at different points depending on their design intent. Complicating things is proximity effect.

Let me think out loud a bit are throw some data points out there and see what floats..

Dipole rolloff is 6db/octave, but the -3dB point in cardioids seems to vary anywhere from around 100 to 500Hz . The Microtech Gefell cardioids I use roll off from about 200Hz down, the MG supercards I use are actually flatter down to a lower frequency than the cardioids. Both have low-cut filters on their amplifier bodies spec'd as "-15dB at 60Hz" but I never engage them.

I pulled up the specification sheet for the Sure KSM 141 at random (since they are good at listing specs), and it has 3 low frequency contour options: Flat, –6 dB/octave below 115 Hz, or –18 dB/octave below 80 Hz. Those are probably intended for proximity compensation and wind-noise reduction.

The DPA 4098 supercardioids I use roll off at 6dB/octave from about 500Hz down. That would seem to preclude them from taper use, but I always use them in arrays in combination with omnis which fill out the bottom end and in all situations they provide a nice clarity without getting bogged down. The bass from them is smoothly attenuated but still there, and can be pulled up using EQ. Actually at bass-heavy events they can sound quite balanced on their own without EQ or the addition of the omnis. Since your SBD feed will be providing bass extension in a somewhat similar way to the omnis in my setup with the 4098, I think in this situation a similar rolloff to the 4098s may be appropriate for your omni Mids. At bit of excess bass rolloff is going to be safer than less given that you have no EQ ability on your mic pair, and since the SBD feed you will be mixing with is presumably full-range.

Given that, I'd probably shoot for -6db/octave from say around 400-500Hz. I think a single pole filter slope in combination with a relatively high corner will blend best with the SBD source. If you didn't have the SBD feed that could be more low-cut than you'd want, but then again it might not as things often get very bass heavy at the front of the stage. For spicing up the SBD with some stereoness, stage-presence, and audience reaction, you really only need the mid-range and higher stuff anyway as long as you have enough bass extension from the SBD feed.

"Given that, I'd probably shoot for -6db/octave from say around 400-500Hz. I think a single pole filter slope in combination with a relatively high corner will blend best with the SBD source. If you didn't have the SBD feed that could be more low-cut than you'd want, but then again it might not as things often get very bass heavy at the front of the stage. For spicing up the SBD with some stereoness, stage-presence, and audience reaction, you really only need the mid-range and higher stuff anyway as long as you have enough bass extension from the SBD feed."

Cool. I learned two* things: (1) dipole (cardioid) rolloff is single pole, good to know; (2) the frequency of the cutoff point varies by make/model. This must be due to some arcane bit of microphone design that I don't know about.

6dB/octave is a simple filter, basically a blocking capacitor in series with the mic output ("outputs" as the mic is balanced so need two caps per mic), loaded into the 2,000 ohm input impedance of the first-gen Scarlett 6i6/18i8 mic preamps, For 500Hz that's a 0.18uF cap, a value easily found in a film type cap for cheap.

I do have to get phantom power out to the mic, and the DC-blocking capacitor won't pass the power from the mic preamp to the mic, so I could either bypass the capacitor with a resistor to leak some current through, which would limit the maximum amount of attenuation at the lowest frequencies, or bang a phantom power supply box on the mic side of the filter. Fortunately I happen to have a dual phantom power supply box. There may even be room in the box to put in the blocking capacitors AND a handy switch to bypass them if they are unneeded.

Yes, best to do the filter after the phantom supply. Double check it in the actual setup after you build it if you can, to confirm the real-world curve ends up. You can then adjust the values if necessary.

Oh, I see. On top of the juggling I'll be doing switching back and forth between two stages with three feeds each, while broadcasting live, wrangling announcers, queuing up and playing station underwriter spots, I also gotta remember to remind youse guys to tune in and listen?

Okay, so I've had two music festivals to tinker with my oddball stage micing idea. Longtime readers may recall that I volunteer for a local community radio station, and each summer we have four music festivals that we broadcast live. When I first started doing this, in 2016, I mainly took a split from the soundboard. The sound guy's responsibility lies with getting good sound in the audience, not giving me a good mix. This has a few problems: first, the feed is mono because sound guys rarely do any panning -- it's not like there is someone in the audience seeking the sweet spot for good imaging; second, loud instruments onstage don't get much love from the PA because they are already loud -- the guitarist with the Marshall stack turned all the way doesn't need any supplementing from the PA, he's plenty loud, so the board feed has little to no guitar. That made the Led Zeppelin tribute band sound like Jimmy Page called in sick. Finally, with only onstage mics, there is no sense that the artists are playing before a live audience.

So what I proposed, and refined with help here, is a modified mid-side mic arrangement, with the mics at the stage lip and up high so as not to block sightlines, and mixed with the mono board feed. I chose to use mid-side not only for its imaging flexibility, but also because it's mono-compatible, and the FM transmitter is not broadcasting a stereo signal.

So -- the results. I am VERY happy with the results. To my ears they sound miles better than the soundboard feed alone.

Nice work. Glad it worked out well. Good life and sparkle to it. This sounds way better than most community radio streams I've heard, which pretty much always sound like the straight SBD. Good example of how the sum of the parts can produce something which supersedes each individual element. Only thing I can't easily check here is the sound of the mono sum as the FM broadcast would have been heard.

Was this an omni MID, used with the high-pass filter you were talking about putting together?

Hi Gutbucket, thanks for all the help. Yes, this was an omni mid mic, with that 500Hz single-pole high pass that you proposed. I added a little more contouring to the mid signal, raised everything from about 2k on up by about 4-5dB as the raw sound was a bit dull in the highs. This mainly to bring out the drum cymbals which frequently get little love from the PA, this added a bit of brilliance to them.

The mono sum is pretty much just the sbd + mid signal, there is no trace of anything the side mic picked up. So, still kind of flat and dimensionless but it does have more going on than just the dry board signal, and the loud instruments that the sound guy didn't include in the PA mix are present.

It's a workable technique. I won't be using it until next June at the earliest, so I've made copious notes and have put the equipment in the closet until then.

I may use a subcard for the mid mic next time, to cut back a bit on the audience chatter and bring a bit more focus to the onstage sound. Maybe change the high pass from 500hz down to 250Hz to compensate for the mic's proximity effect?

I may use a subcard for the mid mic next time, to cut back a bit on the audience chatter and bring a bit more focus to the onstage sound. Maybe change the high pass from 500hz down to 250Hz to compensate for the mic's proximity effect?

I like your EQing of the Mid channel for good brilliance, focus and presence. Choice of Mid pattern will of course determine how much audience reaction is picked up. Need to find the middle way there- not too much or too little, which may depend on the act on stage and time of day. I doubt the Mid mic will be close enough to any one source to get any significant proximity effect, especially if its a subcardioid (proximity effect increases with pattern directivity- most significant with a figure-8, non-existent with an omni, and progressively increases for patterns in between).

What I find myself wondering with regards to the high pass frequency on the Mid is if the tonal balance changes when comparing the stereo against the mono FM version (besides the loss of spatial dimension). The relatively high frequency / low-slope high pass filter on the omni Mid works well for stereo, and is a good complement to the darker side channel contributing heft and nicely increasing stereo width with decreasing frequency. When the side channel is canceled out in mono, I wonder if its not just the stereo spread which collapses but if some of the heft and gravity might be lost as well. A lower high pass filter frequency might correct that if it is an issue.

You may be able to get away with no high pass at all, especially if you move to a more directional Mid. The highpass on the Mid was mostly a safety thing, to make sure you got a usable Mid signal if there were heavy subwoofers or an otherwise overly bass-heavy sound at your stage front Mid/Side mic location, and it doesn't seem sound that was the case.

Thanks for following up and posting the samples. Always cool to hear how these things work out in the real world.

"What I find myself wondering with regards to the high pass frequency on the Mid is if the tonal balance changes when comparing the stereo against the mono FM version (besides the loss of spatial dimension). "

Sure. You could take the MID, SBD, and SIDE stems I shared and mix them yourself, see what you think.

I find that when the monitor output of my DAW is set to mono, the SIDE track contributes little to the mix. What I do gain in mono when mixing the MID to the SBD is a wetter sound, and of course, the PA being the inverse of the stage sound, you can hear the louder instruments that the sound guy didn't need to reinforce much, if at all. Like a Marshall stack, for example. September's festival had a two-guy thrash metal band, and there wasn't much guitar in the PA, so this really helps a lot.

The bigger the band, the more spread out they are, the more this technique shines. Singer/songwriters with mics on voice and guitar (do ANY singer-songwriters play anything other than guitar?) don't gain a whole lot except for ambiance and crowd noise.

"I doubt the Mid mic will be close enough to any one source to get any significant proximity effect, especially if its a subcardioid (proximity effect increases with pattern directivity- most significant with a figure-8, non-existent with an omni, and progressively increases for patterns in between)."

Indeed. The point of the high pass on the omni was to reduce pickup from the subs -- at October's Fall Festival there was a single sub, on the ground at center stage, right where I needed to put my mic stand, so the mic array was about 12 feet directly above the sub. The figure-8 would not pick it up as it is in the pattern's null, but that omni right above it may have gotten slammed. Also, the next street over had another stage with a loud band, and without high-passing the omni, the bass rumble from that stage -- as well as from event generators and the city in general -- was quite audible. Directional mics don't stay flat in the farfield like omnis so there is some built-in high-passing going on with a directional. You need to get in close with a directional mic to get the missing bottom end back in, so yeah, you are right -- I may not need any high-passing if I go directional on the Mid mic. If I used a hypercardioid for the Mid then I reckon random bass (a good name for a band) would not be an issue. But with a subcard I still think I'm going to need to use protection. Bottom-end protection. (I leave this for someone with a dirty mind to comment on.)

I find that when the monitor output of my DAW is set to mono, the SIDE track contributes little to the mix.

That's as it should be, and emulates what is heard in the mono FM feed (at least prior to any other FM signal processing). As the Side signal is routed to left and right channels with opposite polarity in the Mid/Side stereo matrix, when summed to mono in your monitor mix buss it should cancel out, emulating an FM signal which is either mono to begin with (your case) or a weakly received stereo FM signal which looses side-band carrier and reverts to mono. If the Side signal were to be mixed in without using a Mid/Side to Left/Right matrix it wouldn't produce stereo, but the Side signal content also wouldn't cancel out when the stereo signal was summed to mono.

I played your samples directly though a browser on headphones, and was hoping I could find a simple option to sum to mono in the built-in audio output of this computer to check that behavior and any tonality change of the stereo mix, but found no easy option to do so on this machine.

It baffles me that such simple yet immensely useful things have been eliminated from so much of the computer world, where their inclusion requires only a few lines of code. The trend actually seems to be growing worse, driven by over-simplified GUIs designed for phone computing. Just the equivalent of the output section of a 1970's era stereo receiver with a switch for stereo/L+R_mono/L_mono/R_mono and simple bass and treble controls would be great.